MediaPortal Audio renderer - better video playback quality (1 Viewer)

mironicus

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March 9, 2008
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AW: MediaPortal Audio renderer - better video playback quality

Now it would be nice if the realtek drivers were working... Because the ATI drivers are making everytime a HDMI handshake if sound starts/stops... ^^

There is a solution for this problem. Click here and install the ati-anti-pop.exe in your autostart-folder.
 

Truri

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    AW: MediaPortal Audio renderer - better video playback quality

    Yep, thats working, too! :)
    Ok, so i don't need AC3 encode...

    Then I don't have to care what I set in the device properties, extended settings) 16bit/24bit, 48,96kHz etc.?
     

    Truri

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    AW: MediaPortal Audio renderer - better video playback quality

    Now it would be nice if the realtek drivers were working... Because the ATI drivers are making everytime a HDMI handshake if sound starts/stops... ^^

    There is a solution for this problem. Click here and install the ati-anti-pop.exe in your autostart-folder.

    Thank you!
    I will give it a try :D
     

    robyf

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    Re: AW: MediaPortal Audio renderer - better video playback quality

    Bias is only the fps vs. frame rate difference. But there is still that v-sync correction that the audio renderer is able to do. If you enable audio renderer and check with ! stats for the "adjustments done". If that is more than zero, then you would benefit from the audio renderer with live tv as well (it will "remove" the dropped frames on the long run that would be caused by slight drifting of the audio / video clocks.) If that value stays at zero then the other audio render is as good choice :)


    I can confirm this. Without mprenderer I always have a couple of drops in half an hour. With mprenderer I had 0 drops during last night soccer game.
    :D
     

    mironicus

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    March 9, 2008
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    AW: MediaPortal Audio renderer - better video playback quality

    My experience with the MPAudioRenderer is very good. If you watch a Youtube-Video with the plugin "OnlineVideos" the PC won't change the refreshrate before the video starts, so if I use the MPAudioRenderer and a 24fps video is displayed at 50 Hz the video is always smooth without any stutter, without the renderer I'll get constant stutters. :)
     

    tourettes

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    Re: AW: MediaPortal Audio renderer - better video playback quality

    Now it would be nice if the realtek drivers were working... Because the ATI drivers are making everytime a HDMI handshake if sound starts/stops... ^^

    There is a solution for this problem. Click here and install the ati-anti-pop.exe in your autostart-folder.

    Looks like such hack could indeed help. Althou I would prefer the audio drivers to be get fixed...

    The program will automatically drop the audio connection, if exclusive use of the audio device is requested (e.g., bitstream audio, using PAP, change of speaker config, using ReClock WASAPI, ...) and will reconnect if the audio device is available again.
     

    tourettes

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    Re: AW: MediaPortal Audio renderer - better video playback quality

    Ok, so i don't need AC3 encode...

    It will cause following side effects if you use AC3 encoding with HDMI (all limitations are of cource true for SPDIF as well)

    • only 32.0, 44.1 and 48.0 Khz sample rates are working (and audio card must support those, 32 is usually not supported)
    • Only 5.1 channels (HDMI has 8 as max)
    • Lower sound quality - encoding to lossy audio format (AC3)

    Then I don't have to care what I set in the device properties, extended settings) 16bit/24bit, 48,96kHz etc.?

    WASAPI exclusive mode should do that automatically (unless something is broken in drivers). In EXCLUSIVE mode audio renderer has full control over the audio settings.
     

    tourettes

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    Re: AW: MediaPortal Audio renderer - better video playback quality

    Yep, thats working, too! :)

    Nice, since this was coded "blindly" (biggest work was done by arion) as I (nor arion :)) have no access to > 2 channels with HDMI. Log shows as well that it works nicely.

    23-08-2010 15:10:13.045 [ f68] WAVEFORMATEX - WASAPIRenderer::CheckAudioClient
    23-08-2010 15:10:13.045 [ f68] nAvgBytesPerSec 576000
    23-08-2010 15:10:13.045 [ f68] nBlockAlign 12
    23-08-2010 15:10:13.045 [ f68] nChannels 6
    23-08-2010 15:10:13.045 [ f68] nSamplesPerSec 48000
    23-08-2010 15:10:13.045 [ f68] wBitsPerSample 16
    23-08-2010 15:10:13.045 [ f68] wFormatTag 65534
    23-08-2010 15:10:13.045 [ f68] WAVE_FORMAT_EXTENSIBLE
    23-08-2010 15:10:13.045 [ f68] dwChannelMask 63
    23-08-2010 15:10:13.045 [ f68] GUID {00000001-0000-0010-8000-00AA00389B71}
    23-08-2010 15:10:13.045 [ f68] WASAPIRenderer::CheckAudioClient First initialization of the audio renderer
    23-08-2010 15:10:13.045 [ f68] WASAPIRenderer::InitAudioClient
    23-08-2010 15:10:13.046 [ f68] WASAPIRenderer::InitAudioClient format supported
    23-08-2010 15:10:13.067 [ f68] WASAPIRenderer::InitAudioClient service initialization success
    23-08-2010 15:10:13.067 [ f68] WASAPIRenderer::confused:tartAudioClient

    Thanks for confirming.

    :D
     

    tourettes

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    Re: AW: MediaPortal Audio renderer - better video playback quality

    Bias is only the fps vs. frame rate difference. But there is still that v-sync correction that the audio renderer is able to do. If you enable audio renderer and check with ! stats for the "adjustments done". If that is more than zero, then you would benefit from the audio renderer with live tv as well (it will "remove" the dropped frames on the long run that would be caused by slight drifting of the audio / video clocks.) If that value stays at zero then the other audio render is as good choice :)


    I can confirm this. Without mprenderer I always have a couple of drops in half an hour. With mprenderer I had 0 drops during last night soccer game.
    :D

    Yep, that is the benefit of the v-sync correction that EVR presenter and audio renderer can do when they work together.
     

    Truri

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    AW: Re: AW: MediaPortal Audio renderer - better video playback quality

    Ok, so i don't need AC3 encode...

    It will cause following side effects if you use AC3 encoding with HDMI (all limitations are of cource true for SPDIF as well)

    • only 32.0, 44.1 and 48.0 Khz sample rates are working (and audio card must support those, 32 is usually not supported)
    • Only 5.1 channels (HDMI has 8 as max)
    • Lower sound quality - encoding to lossy audio format (AC3)

    Then I don't have to care what I set in the device properties, extended settings) 16bit/24bit, 48,96kHz etc.?

    WASAPI exclusive mode should do that automatically (unless something is broken in drivers). In EXCLUSIVE mode audio renderer has full control over the audio settings.

    Ok, thx for your explanation :)
    Yeah then the Multichannel PCM via HDMI is a nice thing :D

    I will try the renderer in daily use and see whats happening with the various video and audio tracks... :D
    Until now I only "watched" at the sound, so now I will watch at the video too (sync benefits) ^^

    Big thx for your work! Codeing that part blindly... Respect!
    If there is something more specific to test, feel free to ask!

    Hi, again... ;)

    I have a problem playing a mkv.

    Code:
    General
    Complete name                    : I:\[Coalgirls]_Heroic_Age_01-09_(1920x1080_Blu-Ray_FLAC)\[Coalgirls]_Heroic_Age_01_(1920x1080_Blu-Ray_FLAC)_[FA1D197F].mkv
    Format                           : Matroska
    File size                        : 2.88 GiB
    Duration                         : 23mn 55s
    Overall bit rate                 : 17.3 Mbps
    Encoded date                     : UTC 2010-08-17 01:20:29
    Writing application              : mkvmerge v3.4.0 ('Rapunzel') built on May 15 2010 09:38:20
    Writing library                  : libebml v0.8.0 + libmatroska v0.9.0
    
    Video
    ID                               : 1
    Format                           : AVC
    Format/Info                      : Advanced Video Codec
    Format profile                   : High@L5.1
    Format settings, CABAC           : Yes
    Format settings, ReFrames        : 7 frames
    Muxing mode                      : Container profile=Unknown@5.1
    Codec ID                         : V_MPEG4/ISO/AVC
    Duration                         : 23mn 55s
    Width                            : 1 920 pixels
    Height                           : 1 080 pixels
    Display aspect ratio             : 16:9
    Frame rate                       : 23.976 fps
    Color space                      : YUV
    Chroma subsampling               : 4:2:0
    Bit depth                        : 8 bits
    Scan type                        : Progressive
    Title                            : 01 - The Ruined Planet
    Writing library                  : x264 core 96 r1613 81e75e9
    Encoding settings                : cabac=1 / ref=8 / deblock=1:0:0 / analyse=0x3:0x133 / me=esa / subme=9 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=9 / sliced_threads=0 / slices=4 / nr=0 / decimate=1 / interlaced=0 / constrained_intra=0 / bframes=3 / b_pyramid=1 / b_adapt=2 / b_bias=0 / direct=1 / weightb=1 / weightp=2 / keyint=24 / keyint_min=2 / scenecut=40 / intra_refresh=0 / rc_lookahead=24 / rc=crf / mbtree=1 / crf=15.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / vbv_maxrate=40000 / vbv_bufsize=30000 / crf_max=0.0 / ip_ratio=1.40 / aq=1:1.00 / nal_hrd=vbr
    Language                         : English
    
    Audio #1
    ID                               : 2
    Format                           : FLAC
    Format/Info                      : Free Lossless Audio Codec
    Codec ID                         : A_FLAC
    Duration                         : 23mn 55s
    Bit rate mode                    : Variable
    Channel(s)                       : 2 channels
    Sampling rate                    : 48.0 KHz
    Bit depth                        : 24 bits
    Title                            : Japanese 2.0
    Writing library                  : libFLAC 1.2.1 (UTC 2007-09-17)
    Language                         : Japanese
    
    Audio #2
    ID                               : 3
    Format                           : FLAC
    Format/Info                      : Free Lossless Audio Codec
    Codec ID                         : A_FLAC
    Duration                         : 23mn 55s
    Bit rate mode                    : Variable
    Channel(s)                       : 6 channels
    Sampling rate                    : 48.0 KHz
    Bit depth                        : 24 bits
    Title                            : English 5.1
    Writing library                  : libFLAC 1.2.1 (UTC 2007-09-17)
    Language                         : English
    
    Text #1
    ID                               : 4
    Format                           : ASS
    Codec ID                         : S_TEXT/ASS
    Codec ID/Info                    : Advanced Sub Station Alpha
    Language                         : English
    
    Text #2
    ID                               : 5
    Format                           : ASS
    Codec ID                         : S_TEXT/ASS
    Codec ID/Info                    : Advanced Sub Station Alpha
    Title                            : OP/ED
    Language                         : English

    Is that because of the 24bit?!
    The First 2.0 Track plays fine. But switching to the second, 5.1 track, stops playback.
     

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